EP1075768B1 - Method and system for improved call setup - Google Patents
Method and system for improved call setup Download PDFInfo
- Publication number
- EP1075768B1 EP1075768B1 EP99919006A EP99919006A EP1075768B1 EP 1075768 B1 EP1075768 B1 EP 1075768B1 EP 99919006 A EP99919006 A EP 99919006A EP 99919006 A EP99919006 A EP 99919006A EP 1075768 B1 EP1075768 B1 EP 1075768B1
- Authority
- EP
- European Patent Office
- Prior art keywords
- vsp
- call
- message
- called
- calling party
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Expired - Lifetime
Links
Images
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04Q—SELECTING
- H04Q3/00—Selecting arrangements
- H04Q3/0016—Arrangements providing connection between exchanges
- H04Q3/0029—Provisions for intelligent networking
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/50—Centralised arrangements for answering calls; Centralised arrangements for recording messages for absent or busy subscribers ; Centralised arrangements for recording messages
- H04M3/51—Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing
- H04M3/5158—Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing in combination with automated outdialling systems
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/50—Centralised arrangements for answering calls; Centralised arrangements for recording messages for absent or busy subscribers ; Centralised arrangements for recording messages
- H04M3/51—Centralised call answering arrangements requiring operator intervention, e.g. call or contact centers for telemarketing
- H04M3/5183—Call or contact centers with computer-telephony arrangements
- H04M3/5191—Call or contact centers with computer-telephony arrangements interacting with the Internet
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/0024—Services and arrangements where telephone services are combined with data services
- H04M7/003—Click to dial services
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/48—Arrangements for recalling a calling subscriber when the wanted subscriber ceases to be busy
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/06—Arrangements for interconnection between switching centres using auxiliary connections for control or supervision, e.g. where the auxiliary connection is a signalling system number 7 link
Definitions
- This invention relates generally to the completion of voice connections in a switched telephone network, and in particular to the completion of voice connections in a switched telephone network using a data request message for initiating the voice connection and TCAP common channel signaling messages as required to determine the availability of a voice terminal involved in the call.
- Computer telephony integration products are well-known and widely used to provide personal as well as corporate telephone services.
- the use of computer telephony integration products permits enhanced dialling and call handling features.
- a shortcoming of such products is that those features are enabled outside the Public Switched Telephone Network (PSTN). Consequently, call completions effected using such products often use redundant circuits in the PSTN. Efficiency is therefore sacrificed.
- a further disadvantage of such products is that they have no access to the signaling network which controls the PSTN. Consequently, such products are incapable of determining the status of a remote user line or querying PSTN nodes to obtain information useful in call setup or call direction. There therefore exists a need for computer telephony integration products that are more completely integrated with the PSTN.
- call centers which offer customer support, help lines, or the like. Such centers typically use an out-dialer to set up calls to a predetermined list or queue of numbers which are to be called.
- algorithms have been developed to predict when an agent will become available to take a call, and calls are placed in advance of agent availability. If a number dialled is not answered, the number is moved to a bottom of the queue and retried when it has advanced again to the top of the queue. Due to the fact that such dialers have no access to the PSTN signaling network, it is difficult or impossible to write algorithms to accurately and consistently determine the reason that any particular call is not answered when dialled. Consequently, calls may be attempted many times over even though there is no probability of reaching the called party. This wastes transport and signaling facilities and ties up resources that could be profitably used by others. There therefore exists a need for better out-dialer facilities for call centers.
- the system disclosed in applicant's co-pending application includes a Virtual Switching Point (VSP) which is a physical node in the signaling network of the STN and a virtual node in the transmission network of the STN.
- VSP is enabled to receive call request messages from a data network such as a local area network (LAN), a wide area network (WAN), an Intranet or the Internet.
- the VSP processes the call request which may include more than one called number.
- a call request is processed by sending a common channel signaling message from the VSP to an SSP in a local calling area of the calling party to initiate a voice connection with the calling party.
- a second common channel signaling message is sent to a switching point in the STN to initiate a connection with the called party.
- the two connections are the first and second legs to the same call.
- the voice connections may be local or long distance voice connections.
- TCAP Transaction Capability Application Part
- TCAP signaling in a switched telephone network is A METHOD OF ESTABLISHING A COMMUNICATIONS CALL described in PCT/AU57/00163 which was filed on 13 March 1997 and published on 25 September 1997.
- This patent describes a method of establishing a communications call by selecting a called party (6) using an interactive device (16) connected to a public network (10, 12).
- Call address data for the called party (6) is sent along with calling address data for a calling party (4) to a communications platform (18) of the PSTN.
- a call is established between the calling and called parties over the PSTN using the communications platform (18) and the called and calling address data.
- the called address data can be accessed from the public network, and may reside on a server of a messaging network such as the Internet.
- the call is initiated from the communications platform by sending a TCAP message through an IN switch (60) that supports a Service Switch Function (SSF).
- the switch (60) includes an SSF call module (62) to receive and act on connection control signals sent from the communication platform (18).
- the call module (62) invokes Basic Call State Model (BCSM) call leg procedures to contact and connect the called and the calling parties over the PSTN. The availability of neither party is checked before or after the TCAP message is set.
- BCSM Basic Call State Model
- ETU enhanced services unit
- the ETUs (10) serve as local SCPs to provide the Exchanges (20) with enhanced services by receiving ISUP signaling messages for selected calls routed to the ETU (10) by an exchange (20) to which the ETU (10) is connected.
- an ISUP signaling message is received by the ETU (10) a table lookup is triggered.
- the table lookup permits translation of the number to direct the call to an alternate destination depending on triggers in the table lookup.
- TCAP signaling is not used, and no queries are sent from the ETU (10) to the associated exchange (20). Nor does the ETU (10) have any capacity to initiate or control calls aside from redirection of a call initiated elsewhere in the network.
- the service node plays a recorded message selected by the special service subscriber to the caller while the caller waits for a connection to the called special service subscriber line.
- a queue is established in the service control point of all calls placed on hold. While calls are waiting, the calling line is monitored to determined when it becomes idle.
- the SCP (26) initiates the monitoring by sending a monitor-for-change message to the switch (15). When the special service subscriber line becomes idle, a response message is sent to the SCP which selects a waiting call from the queue to be connected by the switch (15) to the called party line. Only subscribers to the special service are treated in this way, and calls disconnected while queued cannot be re-established.
- the ISCP adds a record to the queue which includes the called and calling party numbers, and the time of the call.
- the telephone switching office serving the special service subscriber notifies the ISCP of the availability of the line.
- the ISCP then accesses the queue and notifies the originating telephone switching office to ring both the calling party identified in the queue and the subscriber.
- the telephone switching office serving the subscriber sends a message to the ISCP regarding the incoming call.
- the ISCP recognizing the calling party as the number at the top of the queue sends a command to the telephone switching office to terminate the call to the subscriber, at which point the queue is advanced for the next caller when the subscriber disconnects the call.
- the ISCP (20) After retrieving the next calling party information from the queue, the ISCP (20) sends a TCAP information message to the SSP serving the next calling party to initiate a call-back operation.
- the TCAP information message initiates a "Repeat Call” feature with respect to the line of the special service subscriber.
- the message identifies the destination number for use with the Repeat Call feature. This service is only available for AIN switches that are equipped with inbound call triggers set on special subscriber lines.
- ISUP Integrated Services Digital Network User Part
- VSP Virtual Switching Point
- the invention therefore provides a method of completing a voice connection between a first and second voice terminal on a Switched Telephone Network (STN) (10), including the steps of receiving at a Virtual Switching Point (VSP) (48) a call request message from a calling party (30, 32), the call request message being received through a connection to a data network (34); sending a first Common Channel Signaling (CCS) message from the VSP over a signaling network of the STN to an SSP (18) in the STN to initiate a first leg of the voice connection, and sending a second CCS massage from the VSP to another SSP in the STN to initiate a second leg of the voice connection, the first and second CCS messages being interrelated to an extent that a circuit identification code in each message is associated with opposite ends of the same trunk facility, CHARACTERIZED by:
- the invention further provides a system for using TCAP signaling for improved call set up in a switched telephone network (STN) (10), that includes a virtual switching point (VSP) (48) which is a physical node in a common channel signaling network of the STN and a logical node in a transport network of the STN, a plurality of service switching points (SSPs) (12-18) in the STN having TCAP signaling capability, and a trunk facility (20, 64) in the STN, the VSP being a logical switching node logically located between physical ends of the trunk facility, CHARACTERIZED by:
- STN switched telephone network
- VSP virtual switching point
- SSPs service switching points
- the invention further provides a method of completing calls through a switched telephone network for a call center (70) having a predetermined list of numbers to be called, including the steps of placing the list of numbers to be called in a memory queue which may be accessed by a virtual switching point (VSP) (48) in the network, formulating a first Integrated Services Digital Network User Part (ISUP) signaling Initial Address Message (IAM) at the VSP, the IAM containing a called number from the queue, and forwarding the IAM to initiate a call connection with the called number, CHARACTERIZED by:
- VSP virtual switching point
- ISUP Integrated Services Digital Network User Part
- IAM Initial Address Message
- the invention also provides an out-dialer for a call center (70), CHARACTERIZED by:
- the invention therefore provides a method of completing a voice connection between first and second voice terminals on an STN using TCAP signaling for improved call setup from a VSP.
- the VSP is a physical node in the signaling network of the STN and a logical node in the transport network of the STN.
- the VSP receives call request messages through a data connection.
- the call request messages are requests to set up call connections between a calling party which originates the call request messages and one or more called parties with which the calling party wishes to be connected.
- Call request messages may be initiated by users who log on to a server that may be accessed through a data network.
- the server may be a world-wide web server which is accessed through the Internet.
- Many calling parties have only single line access to the STN and use that access for both data communications and voice communications. Consequently, many calling parties use the same calling line connection to the STN for originating call request messages and for accepting calls set up by the VSP in response to the call request messages.
- the VSP On receipt of a call request message, the VSP is unaware of whether the calling party line has yet become available to accept a call. Consequently, the VSP preferably sends a TCAP message to an SSP in the STN which serves the calling party.
- the TCAP message requests that the SSP monitor the calling party line to determine when it is available to accept calls.
- a TCAP message is returned to the VSP which prompts the VSP to commence call setup in response to the call request message.
- This method can also be used under certain circumstances to efficiently determine the availability of a called party line.
- IAM ISUP Initial Address Message
- the VSP may connect the calling party to an Interactive Voice Response unit (IVR) to obtain information about how the calling party prefers the VSP to proceed.
- IVR Interactive Voice Response unit
- a number of predefined options may be provided to the calling party by the IVR, which permits the calling party to select a preferred option for continuing with the call request. After the calling party indicates a preferred option, the IVR passes the preferred option to the VSP which responds accordingly.
- the VSP may be requested to monitor the called party line during a specific future time interval, the time interval being specified to the IVR by the calling party.
- the method and system in accordance with the invention can also be used to improve call setup for call centers having a plurality of predetermined calls to be completed.
- the system in accordance with the invention permits intelligent call setup for such call centers and eliminates the problems prevalent in the prior art. If a VSP is used for establishing call center connections to predetermined lists of called numbers, transport and signaling facilities are efficiently used.
- the VSP uses the system of the invention when the VSP attempts a call setup to a called party number which returns an ISUP Release (REL) message, the VSP sends a TCAP message to determine a status of the line. If the line is idle, it is assumed that the party is not at the number. The number is therefore unqueued and added to a rejection list to be tried later or on another day.
- REL ISUP Release
- the VSP sends a TCAP message requesting that the called party's SSP perform a line scan to determine when the line is available.
- a TCAP response is returned indicating the called party line is available, the number is queued ahead and the call attempted if an agent is available or predicted to be soon available.
- the method and system in accordance with the invention therefore significantly improves the functionality of the VSP and can significantly reduce the amount of ISUP signaling required to set up call connections.
- This invention relates to a method and system for using TCAP signaling for improved call setup using a virtual signaling point (VSP) in a switched telephone network (STN).
- VSP virtual signaling point
- STN switched telephone network
- Fig. 1 is a schematic diagram of an STN 10 equipped with a system for practising methods in accordance with the invention.
- the STN 10 includes a plurality of switching nodes generally referred to as Service Switching Points (SSP) 12,14,16 and 18 as well as a plurality of other SSPs which are not illustrated.
- SSPs 12-18 are connected to the STN and to each other by trunk groups 20 which are facilities for transporting voice, fax or data.
- the STN includes a signaling network and a transmission network.
- the trunk groups 20 are part of the transmission network.
- a plurality of signaling links make up the signaling network of the STN.
- Each signaling link 22 connects an SSP 12-18 with a Signal Transfer Point (STP) 24,25.
- STP Signal Transfer Point
- the STPs 24,25 are normally arranged in redundant pairs and each SSP 12-18 has a signaling link 22 to each of the STPs 24,25 in the pair.
- the STP pairs 24,25 are also interconnected with signaling links in a manner well known in the art, although for the sake of simplicity those links are not shown in Fig. 1.
- the SSPs 12-18, the STP pairs 24,25 the trunks 20 and the signaling links 22 comprise the basic components of the STN 10.
- An STN 10 will normally include at least one Service Control Point (SCP) which for the sake of simplicity is not illustrated as it is not directly relevant to the invention.
- SCP Service Control Point
- the STN 10 serves a plurality of subscribers having voice terminals 26, 28, 30 and 66.
- the subscribers may also have Personal Computers (PC) 32 connected to the STN by a dial-up connection.
- the PCs 32 are normally used to access the Internet 34 through an Internet Service Provider (ISP) 36,38 or 40.
- ISP Internet Service Provider
- the ISPs 36,38 are connected to SSPs 14,16 by data transmission links 42 which may be a T1 trunk with a modem attached to each end, an Integrated Services Digital Network Basic Rate Interface (ISDN, BRI), a line appearance with a modem attached to each end, a splitter with an Asynchronous Digital Subscriber Loop (ADSL) connection, or the like.
- the ISPs 36,38,40 are connected to the Internet 34 by data transmission links 44 which may be any one of the data links described above, for example.
- the system in accordance with the invention includes a Virtual Switching Point (VSP) 46,48.
- VSP 46,48 is a physical node in the signaling network of the STN and a virtual node in the transmission network of the STN, as will be explained below in some detail.
- VSP 46 is connected to the STP pair 24 by a signaling link 50 and the VSP 48 is connected to STP pair 25 by a signaling link 52.
- the signaling links 50,52 may be A, B or D links, as is well known in the art.
- the VSP 46 is also connected to the ISP 36 by a data link 54 and the VSP 48 is connected to the ISP 38 by data link 56.
- the Internet gateways 55 are deployed on links 54,56 to protect the CCS network from unauthorized use.
- the data links 54,56 may comprise any data transmission link such as a line appearance with a modem on each end, and may use a transmission protocol such as the TCP/IP protocol.
- the system shown in Fig. 1 further includes a public server 58 which may be a World Wide Web (WWW) server well known in the art.
- the public server 58 is connected to the ISP 40 by a data link 60.
- the system may further include an Interactive Voice Recognition Unit (IVR 62) connected to SSP 14 by a trunk 20, typically a DS1 or a T1 facility.
- IVR 62 Interactive Voice Recognition Unit
- the function of the VSPs 46,48; the public server 58 and the IVR 62 will be explained below in some detail.
- the system shown in Fig. 1 is intended to be used by subscribers to the STN 10 having PCs 32 with access to the Internet 34.
- a subscriber with a PC 32 is able to set up call connections by accessing the public server 58. Call requests are entered at the public server 58.
- the public server 58 passes the call requests to a VSP located in a local calling area of the subscriber making the request.
- the subscriber uses PC 32 to make a call request, which is forwarded by the public server 58 to the VSP 48.
- the VSP 48 sends a first signaling message to the SSP 18 to establish a first voice connection with the subscriber telephone 30.
- the VSP 48 sends a second common channel signaling message to an SSP in the STN to initiate a second leg of the voice connection between the subscriber telephone 30 and a called party which may be any one of subscriber telephones 26,28 or 66.
- the SSP which receives the second common channel signaling message depends on the structure of the network and, to a lesser extent, the location of the called party. In order to ensure that signaling messages which permit the VSP 48 to control the call connection are routed through the VSP 48, the call must be routed over facilities which are logically, though not physically, connected to the VSP 48. This may be accomplished in at least one of the two ways.
- a loop-back trunk group 64 connected to SSP 18 may be assigned link sets which are associated with the VSP 48 as if the VSP 48 were a physical switching node in the loop-back trunk group 64.
- one or more members of the trunk groups 20 may be assigned to a trunk group whose link set indicates that the VSP 48 is a switching point in the trunk group, as was explained in detail in applicant's co-pending patent application referenced above.
- the system shown in Fig. 1 further includes a call center 70, typically a call center for telemarketing or customer support which has a mandate to place calls to a predetermined number of called numbers.
- a call center 70 typically includes an automatic call distributor (ACD) or a private branch exchange (PBX) for distributing calls to a number of agents.
- the call center 70 is connected by one or more trunks 72 to SSP 12. Trunk 72 is typically a number of DS0 trunks, an ISDN trunk, or the like.
- a call center 70 equipped in accordance with the invention is also connected by a data communications link 74 to the VSP 46.
- the data communications link 74 need not be a direct connection.
- It may be, for example, a connection through an ISP to the Internet or some other arrangement which permits the VSP 46 and call distribution facilities at the call center 70 to exchange data messages.
- One use of the data connection 74 is to permit the VSP 46 to track the number of agents available and their busy/idle status, as will be explained below in more detail with reference to Fig. 4.
- Fig. 2 shows a schematic illustration of a call flow sequence illustrating the principal actions in a call setup and release sequence in accordance with the invention.
- a calling party using a single line connection to the STN 10 initiates a request by logging on to a public server 58 and completing a form requesting a calling session in which at least one called party number is supplied to the web server 58.
- the web server 58 forwards a call request data message indicated by the dashed line at the top of Fig. 2 over data link 60 to the ISP 40, through the Internet 34, the ISP 38, the gateway 55 and the data link 56 to the VSP 48 (see Fig. 1.).
- VSP 48 On receipt of the call request message, VSP 48 forwards a TCAP query massage over the SS7 signaling network of the STN 10 to SSP 18 requesting that the SSP 18 monitor the calling party line to determine when the calling party line is available to receive a call connection.
- the VSP 48 may automatically send a TCAP query to the calling party's SP.
- the form completed by the calling party using PC 32 may include an indication of whether the calling party has single line access to the STN.
- the TCAP query shown in Fig. 2 is sent only when the form indicates that the calling party has single line access to the STN. If the calling party has multiple line access to the STN, a TCAP query is not sent unless a first call attempt to the calling party's telephone 30 is unsuccessful.
- the TCAP query message sent to the SSP 18 is a Query with Permission (QUERY + P).
- the query message is sent to request that the calling party's SSP monitor the calling party's telephone line to determine when the line becomes idle.
- the SSP 18 will respond with an acknowledgement message in the form of a TCAP message type known as Conversation with Permission (CONVERSATION + P) to indicate to the VSP that the originating switch will perform scanning of the calling line.
- CONVERSATION + P Conversation with Permission
- SSP 18 will return a TCAF RESPONSE message to the VSP 48 to inform the VSP 48 that the calling party's line is available.
- the VSP 48 may use a QUERY + P message to query the busy/idle status of the calling party's line. That query normally returns an immediate TCAP RESPONSE from the SSP 18 with an indication of the busy/idle status of the calling party line.
- the busy/idle status query may be used as an initial inquiry, rather than a request for scanning of the line. The choice of initial queries is dependent on the information available to the VSP, as well as the elapsed time since receipt of the call request message.
- ACM Address Complete
- ANM Answer
- the SSP 18 receives that IAM and assumes that there is an incoming call on the loop-back trunk 64. It examines the called number in the IAM and determines that the caller is served by another SSP in the STN 10, SSP 14 for example.
- the SSP 18 therefore formulates an appropriate IAM and transmits it into the signaling network of the STN 10.
- STN 10 forwards the IAM to the SSP 14 which checks the availability of the called party line.
- a disconnect is received from the calling party 30.
- the SSP 14 returns an RLC message to the SSP 18. Thereafter it applies dial tone to the telephone line of user 26 which responds by going on-hook and the call series is completed.
- the call request initiated by the calling party using PC 32 may have included several called numbers in which case the calling party would normally not be disconnected from the SSP 18 and the VSP on receipt of a release from the called party's SSP would proceed with setting up a call connection to the next called number.
- Figs. 3a and 3b show a portion of a call message flow involved in the handling of a call request initiated by a calling party at PC 32.
- the calling party using PC 32 accesses web server 58 and initiates a call request which is sent through the Internet to the web server 58 and forwarded through the Internet to the VSP 48.
- the call request indicates that the calling party using PC 32 has single line access to the STN 10.
- the VSP 48 therefore forwards a TCAP QUERY + P message to the SSP 18 requesting that the SSP 18 scan the calling party's telephone line to determine when the line becomes available to receive a call.
- the SSP 18 responds with an acknowledgement message, as described above, followed when the calling line becomes available with a TCAP RESPONSE message indicating that the calling line is available to receive a call. As also described above, this prompts the VSP 48 to initiate a call setup in which the VSP 48 connects a first leg of the call connection with the calling party at telephone 30 and then initiates a second leg of the call connection by sending an IAM through the STN 10 to the SSP 14.
- the called line 26 is busy and the SSP 14 returns a release message to the SSP 18.
- the SSP 14 also returns an RLC message to the SSP 14 through the STN 10.
- the SSP 18 determines that the IVR 62 is served by another SSP in the STN 10. It therefore forwards an IAM through the STN 10 to the SSP 14 which serves the IVR 62.
- Ensuing ACM and ANM messages are exchanged, as illustrated. Those messages follow the same sequence as the ACM and ANM described with reference to the top half of the diagram.
- the VSP After sending the IAM to connect the called party telephone 30 to the IVR 62, the VSP also sends a data message to the IVR 62 informing the IVR 62 of the call and requesting that the IVR 62 begin an interactive session with the calling party telephone 30 in order to get instructions as to how the balance of the call request session is to be handled.
- the IVR plays a pre-recorded message to the calling party at telephone 30 requesting that the calling party select an option regarding how the calling session should proceed.
- a few of the possible options which may be enabled are as follows:
- the call is simply abandoned and control is returned to the VSP which attempts to complete the call connection between the calling party and the next called party umber in the call request message.
- a TCAP query is sent by the VSP 48 to the SSP which serves the calling party requesting that the SSP scan the called party line and advise the VSP as soon as the called party line becomes available to accept the call.
- the call is held by the IVR.
- the IVR 62 preferably plays auditory content to the caller while the caller is held.
- the auditory content may be any desirable content including recorded music, a radio station, advertisements, or interactive entertainment which the IVR 62 is programmed to provide.
- the VSP 48 disconnects the calling party line from the IVR 62 and sends a query message to the SSP 14 to monitor the called party line.
- the SSP 14 returns a RESPONSE message indicating that the called party line is available, the VSP attempts to create a call connection between the calling party line and the called party line as described above with reference to Fig. 2.
- the call attempt is abandoned and the VSP attempts to establish a call connection between the calling party and the next called party in the call request message.
- the VSP simultaneously sends a TCAP QUERY + P message to the SSP 14 through the STN 10 requesting that the SSP 14 monitor the called party line and notify the VSP 48 when the called party line becomes available to receive a call.
- the VSP 48 receives a RESPONSE message from the SSP 14 indicating that the called party line is available, the VSP 48 preferably formulates a first IAM to establish a connection with the calling party telephone 30 followed immediately by a second IAM to establish a connection with the called party telephone 26, without waiting for the receipt of ACM and ANM messages. Since the calling party is known to have a call waiting feature, it is assumed that the calling party will respond to the call. Since the VSP 48 has just been notified that the called party line is available, there is an excellent probability that the called party will answer the call.
- option e the user must specify a time interval when the VSP 48 should monitor the called party line for availability. This option would normally be used by a calling party which expects to be available during a certain period of the day, say from 2:00-4:00 p.m.
- option e returned from the IVR 62
- the VSP 48 queues a monitoring request.
- the VSP 48 sends a TCAP QUERY + P message to the SSP 14 requesting that it report the line status of the called party.
- the VSP 48 If the SSP 14 reports that the called party line is busy, the VSP 48 returns a TCAP QUERY + P message requesting that the SSP 14 scan the called party line and advise the VSP 48 when the line becomes available. If the SSP 14 responds before 4:00 p.m. that the calling party line is available, the VSP 48 responds by formulating a first IAM which initiates a call request to the calling party line. After the calling party answers, the VSP 48 initiates a second IAM to complete a second leg of the call connection as described with reference to Fig. 1.
- the calling party after the IVR 62 plays the called party busy options to the calling party at telephone 30, the calling party returns DTMF signals by keying a selection which indicates option b) because the calling party 30 urgently wishes to reach the called party at telephone 26.
- Those instructions are forwarded via a data connection from the IVR 62 to the VSP 48.
- the VSP 48 On receipt of the instructions, the VSP 48 immediately launches a TCAP QUERY + P message to the SSP 14 through the STN 10 requesting that the SSP 14 monitor the line of telephone 26.
- the SSP 14 returns an acknowledgement through the STN 10 and begins a scan of the called party line.
- the SSP 14 returns a TCAP RESPONSE message through the STN 10 advising the VSP 48 that the calling party line is available.
- the SSP 18 formulates an IAM which it forwards through the STN 10 to the SSP 14.
- Fig. 4 shows an exemplary call flow sequence in which VSP 46 (see Fig. 1) is used as an out-dialer to set up calls for a call center 70. Because of the flexibility offered by the VSP, VSP 46 can perform all call setup functions for the call center 70 and because of its position in the signaling network, the VSP 46 is enabled to use PSTN resources much more efficiently than most call center equipment in use today.
- call center 70 sends a called number list to the VSP 46 via a data connection, for example the data communications link 74 (see Fig. 1).
- the called number list may be supplied at any time and may augment a called number queue already resident on VSP 46.
- a called number on the called number list is telephone 66 served by SSP 18 (see Fig. 1).
- the SSP 18 checks the availability of telephone 66 and determines that it is busy.
- the SSP 18 therefore returns an ISUP Release (REL) message indicating to the VSP 46 that the call cannot be completed.
- REL ISUP Release
- RLC Release Complete
- the VSP In order to determine the status of telephone 66, the VSP sends a TCAP Query with Permission (Q+P) message to the SSP 18 requesting that the SSP 18 monitor the called party line and advise when the line becomes idle. SSP 18 responds to the TCAP message with a TCAP Conversation with Permission (C+P) shown in the diagram as an "ACK" response.
- Q+P Permission
- C+P TCAP Conversation with Permission
- the SSP 18 sends a TCAP response message to the VSP 46 reporting that the called party line is idle.
- VSP 46 responds to the message by sending a data query message to the call center 70 to determine whether an agent is available to take the call. Alternatively, the VSP may maintain an agent status file in which it maintains the current status of agents that is routinely updated by the call center, or the VSP 46.
- the call center 70 responds that there is an agent available to take the call. Consequently, the VSP 46 launches an IAM with a Circuit Identification Code (CIC) of "001" to connect with the called party.
- CIC Circuit Identification Code
- the SSP 18 therefore formulates an IAM which is forwarded through the STN 10 to the SSP 12.
- SSP 12 forwards the IAM to the call center 70 which in this example is assumed to be served by an ISDN trunk with PRI signaling.
- the ACD or PBX at the call center returns an ACM indicating that an agent phone is available to take the call and an ACM message is sent by SSP 12 back to the SSP 18 which returns an ACM to the VSP 46.
- the call center applies a ringing tone or some other advisement message to the agent at telephone 76.
- the ACD or PBX at the call center returns an ANM message to the SSP 12 which sends an ANM message to the SSP 18.
- agent efficiency can be further improved, while call control is maintained.
- Fig. 5 further illustrates the use of a VSP 46 as an out-dialer for call center 70.
- VSP 46 illustrates efficient use of the STN enabled by the VSP 46 when configured to function as an out-dialer for the call center 70.
- the called number is to a number served by the SSP 18 which is no longer in service.
- the SSP 18 responds by returning an ISUP REL message with a cause indicating that the number is unavailable.
- the VSP 46 may log such numbers and include them in an exception report which is periodically sent over the data communications link 74 to the call center 70 or reported in some other manner.
- an IAM including a new called number is formulated by the VSP 46.
- PAM Pass Along Message
- VSP 46 may query the call center 70, or a list the VSP 46 maintains, to determine if an agent is available to take the call. The VSP 46 completes the call to the called party if the agent is available, as explained above with reference to Fig. 4. Alternatively, the VSP 46 may add the call to the head of the memory queue of numbers to be called. If, however, the TCAP response received by the SSP 18 indicates that the called party line is idle rather than busy, the VSP 46 is enabled to deduce that the called party line is unattended. The called number may therefore be placed in a special queue to be called at a later time. Algorithms may be written to determine when calls placed in a special queue are to be retried.
- the call may be placed at the bottom of the memory queue of numbers to be called and retried when it arrives again at ahead of the queue.
- a call attempt count associated with the number may be incremented when the number is placed at the bottom of the queue. If this option is used, every time a call attempt is made using a call from the memory queue of numbers to be called, the call attempt count is examined to determine whether it has exceeded a predetermined limit, and the called number is removed from the list and added to an exception report, or the like, if the call attempt count exceeds some predetermined value.
- the VSP 46 therefore provides an excellent out-dialer for a call center which permits calls to be placed much more economically and intelligently. It also provides an economical mechanism for removing unavailable numbers from called number lists and for efficiently and accurately handling calls to numbers where no called party is available.
- the system and the method in accordance with the invention therefore provide additional functionality in the STN which permits service providers to offer more options to calling parties wishing to automate call completion. It also minimizes common channel signaling by ensuring that call setup messages are sent at opportune times to ensure with reasonable certainty that call completion will ensue.
Landscapes
- Engineering & Computer Science (AREA)
- Signal Processing (AREA)
- Business, Economics & Management (AREA)
- Marketing (AREA)
- Computer Networks & Wireless Communication (AREA)
- General Engineering & Computer Science (AREA)
- Telephonic Communication Services (AREA)
- Exchange Systems With Centralized Control (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
- Mechanical Coupling Of Light Guides (AREA)
- Time-Division Multiplex Systems (AREA)
Abstract
Description
- This invention relates generally to the completion of voice connections in a switched telephone network, and in particular to the completion of voice connections in a switched telephone network using a data request message for initiating the voice connection and TCAP common channel signaling messages as required to determine the availability of a voice terminal involved in the call.
- Computer telephony integration products are well-known and widely used to provide personal as well as corporate telephone services. The use of computer telephony integration products permits enhanced dialling and call handling features. A shortcoming of such products is that those features are enabled outside the Public Switched Telephone Network (PSTN). Consequently, call completions effected using such products often use redundant circuits in the PSTN. Efficiency is therefore sacrificed. A further disadvantage of such products is that they have no access to the signaling network which controls the PSTN. Consequently, such products are incapable of determining the status of a remote user line or querying PSTN nodes to obtain information useful in call setup or call direction. There therefore exists a need for computer telephony integration products that are more completely integrated with the PSTN.
- Another commonly used system in the PSTN are call centers which offer customer support, help lines, or the like. Such centers typically use an out-dialer to set up calls to a predetermined list or queue of numbers which are to be called. To improve performance, algorithms have been developed to predict when an agent will become available to take a call, and calls are placed in advance of agent availability. If a number dialled is not answered, the number is moved to a bottom of the queue and retried when it has advanced again to the top of the queue. Due to the fact that such dialers have no access to the PSTN signaling network, it is difficult or impossible to write algorithms to accurately and consistently determine the reason that any particular call is not answered when dialled. Consequently, calls may be attempted many times over even though there is no probability of reaching the called party. This wastes transport and signaling facilities and ties up resources that could be profitably used by others. There therefore exists a need for better out-dialer facilities for call centers.
- In applicant's co-pending PCT patent application No. PCT/CA98/01191 filed 22 December 1998, a method and a system for completing voice connections between voice terminals in a Switched Telephone Network (STN) using the flexibility of computer control exercised through a data network independently of voice terminals connected to the network was disclosed. The method and system provide several advantages over prior art methods of completing voice connections. First, it provides all the advantages and flexibility of computer control, including automated dialling from electronic telephone books or directories. It also provides the advantage of sequential calling without disconnection of the calling party so that a plurality of sequential calls may be completed without interrupting the user's voice connection with an originating service switching point (SSP) in the STN.
- The system disclosed in applicant's co-pending application includes a Virtual Switching Point (VSP) which is a physical node in the signaling network of the STN and a virtual node in the transmission network of the STN. The VSP is enabled to receive call request messages from a data network such as a local area network (LAN), a wide area network (WAN), an Intranet or the Internet. The VSP processes the call request which may include more than one called number. A call request is processed by sending a common channel signaling message from the VSP to an SSP in a local calling area of the calling party to initiate a voice connection with the calling party. After the connection is established with the calling party, a second common channel signaling message is sent to a switching point in the STN to initiate a connection with the called party. The two connections are the first and second legs to the same call. The voice connections may be local or long distance voice connections.
- While this system provides significantly improved functionality over the computer-integrated telephony systems of the prior art, and capitalizes on the inherent switching capability of the PSTN, it does not capitalize on the inherent query capability of a common channel signaling network. In particular, the common channel signaling system, Signaling System 7 (SS7) provides a guery signaling capability known as Transaction Capability Application Part (TCAP) signaling. TCAP signaling provides a powerful query tool which may be used by physical nodes in an SS7 network.
- Exemplary of the use of TCAP signaling in a switched telephone network is A METHOD OF ESTABLISHING A COMMUNICATIONS CALL described in PCT/AU57/00163 which was filed on 13 March 1997 and published on 25 September 1997. This patent describes a method of establishing a communications call by selecting a called party (6) using an interactive device (16) connected to a public network (10, 12). Call address data for the called party (6) is sent along with calling address data for a calling party (4) to a communications platform (18) of the PSTN. A call is established between the calling and called parties over the PSTN using the communications platform (18) and the called and calling address data. The called address data can be accessed from the public network, and may reside on a server of a messaging network such as the Internet. The call is initiated from the communications platform by sending a TCAP message through an IN switch (60) that supports a Service Switch Function (SSF). The switch (60) includes an SSF call module (62) to receive and act on connection control signals sent from the communication platform (18). The call module (62) invokes Basic Call State Model (BCSM) call leg procedures to contact and connect the called and the calling parties over the PSTN. The availability of neither party is checked before or after the TCAP message is set.
- United States Patent No. 5,519,770 which issued on May 21, 1996 to Stein entitled ENHANCED TELEPHONY APPARATUS AND SYSTEM describes an enhanced services unit (ETU) 10 that is connected to existing telephone exchanges (20) by fixed signaling links (24). The ETUs (10) serve as local SCPs to provide the Exchanges (20) with enhanced services by receiving ISUP signaling messages for selected calls routed to the ETU (10) by an exchange (20) to which the ETU (10) is connected. when an ISUP signaling message is received by the ETU (10) a table lookup is triggered. The table lookup permits translation of the number to direct the call to an alternate destination depending on triggers in the table lookup. TCAP signaling is not used, and no queries are sent from the ETU (10) to the associated exchange (20). Nor does the ETU (10) have any capacity to initiate or control calls aside from redirection of a call initiated elsewhere in the network.
- united States Patent No. 5,600,710 which issued on February 4, 1997 to Weisser, Jr. et al, is entitled METHOD FOR PROVIDING A RECORDED MESSAGE TO A TELEPHONE CALLER WHEN A CALLED NUMBER IS BUSY. The patent describes a method in which a call directed to a subscriber of an "Advertised on Busy" service is completed to an SSP serving the subscriber to the service. It is then determined whether the line for the called subscriber is busy. If the called line is not busy, the telephone switch connects the call to the called line. If the called line of the special service subscriber is busy, a query is sent to a service control point (26) and then temporarily connected to a voice circuit of a chosen service node (39). The service node plays a recorded message selected by the special service subscriber to the caller while the caller waits for a connection to the called special service subscriber line. A queue is established in the service control point of all calls placed on hold. While calls are waiting, the calling line is monitored to determined when it becomes idle. The SCP (26) initiates the monitoring by sending a monitor-for-change message to the switch (15). When the special service subscriber line becomes idle, a response message is sent to the SCP which selects a waiting call from the queue to be connected by the switch (15) to the called party line. Only subscribers to the special service are treated in this way, and calls disconnected while queued cannot be re-established.
- United States Patent No. 5,692,033 to Farris entitled AIN QUEUING FOR CALL-SACK SYSTEM issued on November 25, 1997. This patent describes an arrangement in which subscribers to a special service by which a call placed to the special service subscriber's number triggers a query from a telephone switching office serving the subscriber to an integrated services control point (SCP). The SCP instructs the telephone switching office to route the call to an announcement platform that notifies the calling party of the queuing arrangement. If the calling party desires to be entered into the queue, the calling party enters specific digits in response to prompts from an announcement platform. The telephone switching office sends a message to the ISCP to add the calling party to the queue. The ISCP adds a record to the queue which includes the called and calling party numbers, and the time of the call. When the special service subscriber line becomes available, the telephone switching office serving the special service subscriber notifies the ISCP of the availability of the line. The ISCP then accesses the queue and notifies the originating telephone switching office to ring both the calling party identified in the queue and the subscriber. On receiving the incoming call, the telephone switching office serving the subscriber sends a message to the ISCP regarding the incoming call. The ISCP recognizing the calling party as the number at the top of the queue sends a command to the telephone switching office to terminate the call to the subscriber, at which point the queue is advanced for the next caller when the subscriber disconnects the call. After retrieving the next calling party information from the queue, the ISCP (20) sends a TCAP information message to the SSP serving the next calling party to initiate a call-back operation. The TCAP information message initiates a "Repeat Call" feature with respect to the line of the special service subscriber. The message identifies the destination number for use with the Repeat Call feature. This service is only available for AIN switches that are equipped with inbound call triggers set on special subscriber lines.
- There therefore exists a need for call completion systems which use the inherent switching power resident in the PSTN as well as the inherent query capability resident in the common channel signaling network that controls the PSTN.
- It is an object of the invention to provide a method for using TCAP signaling for improved call setup from a Virtual Switching Point in a Switched Telephone Network.
- It is a further object of the invention to provide a method using TCAP signaling massages which originate from a signaling node in the STN for improved call setup.
- it is another object of the invention to provide a method for using TCAP signaling messages in an STN to determine the availability of a calling party's voice terminal in order to minimize Integrated Services Digital Network User Part (ISUP) signaling during initial call setup.
- It is yet a further object of the invention to provide a system for using TCAP signaling for improved call setup from a Virtual Switching Point (VSP) in an STN in which the VSP is a physical node in the signaling network and a virtual node in the transmission network: of the STN.
- It is another object of the invention to provide a system for using TCAP signaling for improved call setup from a VSP wherein the TCAP signaling is used to determine the availability of a called voice terminal when a first attempt to call that voice terminal indicates that the voice terminal is busy.
- It is yet another object of the invention to provide a system for using TCAP signaling for improved call setup for a call center where a plurality of predetermined call requests must be completed.
- It is a further object of the invention to provide an out-dialer for a call center, the out-dialer comprising a VSP enabled with TCAP messaging capability.
- The invention therefore provides a method of completing a voice connection between a first and second voice terminal on a Switched Telephone Network (STN) (10), including the steps of receiving at a Virtual Switching Point (VSP) (48) a call request message from a calling party (30, 32), the call request message being received through a connection to a data network (34); sending a first Common Channel Signaling (CCS) message from the VSP over a signaling network of the STN to an SSP (18) in the STN to initiate a first leg of the voice connection, and sending a second CCS massage from the VSP to another SSP in the STN to initiate a second leg of the voice connection, the first and second CCS messages being interrelated to an extent that a circuit identification code in each message is associated with opposite ends of the same trunk facility, CHARACTERIZED by:
- sending a Transactions Capability Application part (TCAP) query message from the VSP (48) to the SSP (18) that serves the calling party to determine the availability of a calling line identified in the call request message, and initiating a call from the VSP using Integrated Services Digital Network User Part (ISUP) common channel signaling messages sent from the VSP to initiate the first and second legs of the voice connection, if a response to the TCAP query message indicates that the calling line is available.
-
- The invention further provides a system for using TCAP signaling for improved call set up in a switched telephone network (STN) (10), that includes a virtual switching point (VSP) (48) which is a physical node in a common channel signaling network of the STN and a logical node in a transport network of the STN, a plurality of service switching points (SSPs) (12-18) in the STN having TCAP signaling capability, and a trunk facility (20, 64) in the STN, the VSP being a logical switching node logically located between physical ends of the trunk facility, CHARACTERIZED by:
- a server (58) connected to a data network (34) and accessible through the data network, the server accepting call requests from calling parties having access to the data network;
- the VSP (48) is enabled with Transactions Capability Application Part (TCAP) signaling capability and the VSP has a connection to the data network and receives the call requests from the server, the VSP setting up call connections between the calling party and a called party specified in the call request by a called party number by sending a first Integrated Services Digital Network User Part (ISUP) signaling message to the SSP (18) serving the calling party, and another ISUP signaling message to the SSP (12) serving the called party so that the call setup between the called party and the calling party is routed over the trunk facility; and
- a gateway switch located between the VSP and the data network to isolate the VSP from the data network to prevent direct access to the VSP form the data network.
-
- The invention further provides a method of completing calls through a switched telephone network for a call center (70) having a predetermined list of numbers to be called, including the steps of placing the list of numbers to be called in a memory queue which may be accessed by a virtual switching point (VSP) (48) in the network, formulating a first Integrated Services Digital Network User Part (ISUP) signaling Initial Address Message (IAM) at the VSP, the IAM containing a called number from the queue, and forwarding the IAM to initiate a call connection with the called number, CHARACTERIZED by:
- on receipt of an ISUP Address Complete (ACM) message and an ISUP Answer (ANM) message in response to the first IAM, formulating a second IAM at the VSP (48), the second TAM including a number of the call center (70) as the called number and forwarding the second IAM to connect the called number from the queue with the call center, the first and second ISUP IAM messages being related to an extent that the call is routed over a trunk facility in the switched telephone network in which the VSP is a logical switching mode located between opposite ends of the trunk facility.
-
- The invention also provides an out-dialer for a call center (70), CHARACTERIZED by:
- a virtual switching point (VSP) (46) which is a physical node in a signaling network of a switched telephone network (STN) (10) that serves the call center (70), and a virtual node in at least one trunk facility of the STN;
- a memory queue accessible by the VSP, the memory queue containing at least one list of called numbers to which calls are to be set up by the VSP; and
- the VSP (46) sets up calls to called numbers by sending a first ISUP IAM message to a switching point associated with an end of the trunk facility in which the VSP is the virtuel node, the first ISUP IAM message containing the called number; and
- the VSP (46) completes the set up of the call by sending a second ISUP LAM message to a switching point associated with an opposite end of the trunk facility in which the VSP is the virtual node, the second IAM including a called number that is a number of the call center (70).
-
- The invention therefore provides a method of completing a voice connection between first and second voice terminals on an STN using TCAP signaling for improved call setup from a VSP. The VSP is a physical node in the signaling network of the STN and a logical node in the transport network of the STN. The VSP receives call request messages through a data connection. The call request messages are requests to set up call connections between a calling party which originates the call request messages and one or more called parties with which the calling party wishes to be connected.
- Call request messages may be initiated by users who log on to a server that may be accessed through a data network. The server may be a world-wide web server which is accessed through the Internet. Many calling parties have only single line access to the STN and use that access for both data communications and voice communications. Consequently, many calling parties use the same calling line connection to the STN for originating call request messages and for accepting calls set up by the VSP in response to the call request messages. On receipt of a call request message, the VSP is unaware of whether the calling party line has yet become available to accept a call. Consequently, the VSP preferably sends a TCAP message to an SSP in the STN which serves the calling party. The TCAP message requests that the SSP monitor the calling party line to determine when it is available to accept calls. When the calling party line becomes available, a TCAP message is returned to the VSP which prompts the VSP to commence call setup in response to the call request message. Thus unnecessary ISUP messaging is avoided, and the STN's common channel signaling system is efficiently used.
- This method can also be used under certain circumstances to efficiently determine the availability of a called party line. In normal call processing, it is most efficient to send an ISUP Initial Address Message (IAM) to initiate a call connection with the called party. If the called party station is busy, a Release message is returned in response to the ISUP IAM message. When a Release message is received in response to an IAM, and the calling party has a single line connection to the STN, the VSP may connect the calling party to an Interactive Voice Response unit (IVR) to obtain information about how the calling party prefers the VSP to proceed. A number of predefined options may be provided to the calling party by the IVR, which permits the calling party to select a preferred option for continuing with the call request. After the calling party indicates a preferred option, the IVR passes the preferred option to the VSP which responds accordingly.
- Among the preferred options that may be presented are options which instruct the VSP to monitor the called party line. The monitoring is accomplished using a TCAP query message sent to an SSP which serves the called party line. The TCAP query message requests that the SSP monitor the called party line and advise the VSP when the called party line becomes available to accept a call. In accordance with another option, the VSP may be requested to monitor the called party line during a specific future time interval, the time interval being specified to the IVR by the calling party.
- The method and system in accordance with the invention can also be used to improve call setup for call centers having a plurality of predetermined calls to be completed.
- The system in accordance with the invention permits intelligent call setup for such call centers and eliminates the problems prevalent in the prior art. If a VSP is used for establishing call center connections to predetermined lists of called numbers, transport and signaling facilities are efficiently used. Using the system of the invention when the VSP attempts a call setup to a called party number which returns an ISUP Release (REL) message, the VSP sends a TCAP message to determine a status of the line. If the line is idle, it is assumed that the party is not at the number. The number is therefore unqueued and added to a rejection list to be tried later or on another day. If the status returns a "busy" condition, the VSP sends a TCAP message requesting that the called party's SSP perform a line scan to determine when the line is available. When a TCAP response is returned indicating the called party line is available, the number is queued ahead and the call attempted if an agent is available or predicted to be soon available.
- The method and system in accordance with the invention therefore significantly improves the functionality of the VSP and can significantly reduce the amount of ISUP signaling required to set up call connections.
- The invention will now be further explained by way of example only and with reference to the following drawings wherein:
- Fig. 1 is a schematic diagram of a switched telephone network equipped with a system in accordance with the invention for using TCAP signaling for improved call setup from a virtual switching point;
- Fig. 2 is an exemplary call flow sequence showing principal actions in a call setup and release using the method and system in accordance with the invention;
- Figs. 3a and 3b show an exemplary call flow illustrating showing principal actions in call setup in accordance with another use of the method and system in accordance with the invention;.
- Fig. 4 shows an exemplary call flow sequence illustrating principal actions in a call setup sequence using a system in accordance with the invention to set up calls for a call center; and
- Fig. 5 shows exemplary call flow sequences illustrating principal actions in a call setup sequence using a system in accordance with the invention to set up calls for a call center when the called number is not available or the called party line is in use.
-
- This invention relates to a method and system for using TCAP signaling for improved call setup using a virtual signaling point (VSP) in a switched telephone network (STN).
- Fig. 1 is a schematic diagram of an
STN 10 equipped with a system for practising methods in accordance with the invention. TheSTN 10 includes a plurality of switching nodes generally referred to as Service Switching Points (SSP) 12,14,16 and 18 as well as a plurality of other SSPs which are not illustrated. The SSPs 12-18 are connected to the STN and to each other bytrunk groups 20 which are facilities for transporting voice, fax or data. The STN includes a signaling network and a transmission network. The trunk groups 20 are part of the transmission network. A plurality of signaling links make up the signaling network of the STN. Eachsignaling link 22 connects an SSP 12-18 with a Signal Transfer Point (STP) 24,25. TheSTPs signaling link 22 to each of theSTPs - The SSPs 12-18, the STP pairs 24,25 the
trunks 20 and the signaling links 22 comprise the basic components of theSTN 10. AnSTN 10 will normally include at least one Service Control Point (SCP) which for the sake of simplicity is not illustrated as it is not directly relevant to the invention. TheSTN 10 serves a plurality of subscribers havingvoice terminals PCs 32 are normally used to access theInternet 34 through an Internet Service Provider (ISP) 36,38 or 40. TheISPs SSPs data transmission links 42 which may be a T1 trunk with a modem attached to each end, an Integrated Services Digital Network Basic Rate Interface (ISDN, BRI), a line appearance with a modem attached to each end, a splitter with an Asynchronous Digital Subscriber Loop (ADSL) connection, or the like. TheISPs Internet 34 bydata transmission links 44 which may be any one of the data links described above, for example. - The system in accordance with the invention includes a Virtual Switching Point (VSP) 46,48. Each
VSP VSP 46 is connected to theSTP pair 24 by asignaling link 50 and theVSP 48 is connected toSTP pair 25 by asignaling link 52. The signaling links 50,52 may be A, B or D links, as is well known in the art. TheVSP 46 is also connected to theISP 36 by adata link 54 and theVSP 48 is connected to theISP 38 bydata link 56.Internet gateways 55 are deployed onlinks - The system shown in Fig. 1 further includes a
public server 58 which may be a World Wide Web (WWW) server well known in the art. Thepublic server 58 is connected to theISP 40 by adata link 60. The system may further include an Interactive Voice Recognition Unit (IVR 62) connected toSSP 14 by atrunk 20, typically a DS1 or a T1 facility. The function of theVSPs public server 58 and theIVR 62 will be explained below in some detail. - The system shown in Fig. 1 is intended to be used by subscribers to the
STN 10 havingPCs 32 with access to theInternet 34. Using the system in accordance with the invention, a subscriber with aPC 32 is able to set up call connections by accessing thepublic server 58. Call requests are entered at thepublic server 58. Thepublic server 58 passes the call requests to a VSP located in a local calling area of the subscriber making the request. In the example which follows, the subscriber usesPC 32 to make a call request, which is forwarded by thepublic server 58 to theVSP 48. As will be explained below in more detail, on receipt of the call request, theVSP 48 sends a first signaling message to theSSP 18 to establish a first voice connection with thesubscriber telephone 30. After theSSP 18 confirms a connection with thesubscriber telephone 30, theVSP 48 sends a second common channel signaling message to an SSP in the STN to initiate a second leg of the voice connection between thesubscriber telephone 30 and a called party which may be any one ofsubscriber telephones VSP 48 to control the call connection are routed through theVSP 48, the call must be routed over facilities which are logically, though not physically, connected to theVSP 48. This may be accomplished in at least one of the two ways. For example, a loop-back trunk group 64 connected toSSP 18 may be assigned link sets which are associated with theVSP 48 as if theVSP 48 were a physical switching node in the loop-back trunk group 64. Likewise, one or more members of thetrunk groups 20 may be assigned to a trunk group whose link set indicates that theVSP 48 is a switching point in the trunk group, as was explained in detail in applicant's co-pending patent application referenced above. - The system shown in Fig. 1 further includes a
call center 70, typically a call center for telemarketing or customer support which has a mandate to place calls to a predetermined number of called numbers. Such centers typically include an automatic call distributor (ACD) or a private branch exchange (PBX) for distributing calls to a number of agents. Thecall center 70 is connected by one ormore trunks 72 toSSP 12.Trunk 72 is typically a number of DS0 trunks, an ISDN trunk, or the like. Acall center 70 equipped in accordance with the invention is also connected by a data communications link 74 to theVSP 46. The data communications link 74 need not be a direct connection. It may be, for example, a connection through an ISP to the Internet or some other arrangement which permits theVSP 46 and call distribution facilities at thecall center 70 to exchange data messages. One use of thedata connection 74 is to permit theVSP 46 to track the number of agents available and their busy/idle status, as will be explained below in more detail with reference to Fig. 4. - Fig. 2 shows a schematic illustration of a call flow sequence illustrating the principal actions in a call setup and release sequence in accordance with the invention. As shown in Fig. 2, a calling party using a single line connection to the
STN 10 initiates a request by logging on to apublic server 58 and completing a form requesting a calling session in which at least one called party number is supplied to theweb server 58. After the callingparty using PC 32 completes the form onweb server 58, theweb server 58 forwards a call request data message indicated by the dashed line at the top of Fig. 2 over data link 60 to theISP 40, through theInternet 34, theISP 38, thegateway 55 and the data link 56 to the VSP 48 (see Fig. 1.). On receipt of the call request message,VSP 48 forwards a TCAP query massage over the SS7 signaling network of theSTN 10 toSSP 18 requesting that theSSP 18 monitor the calling party line to determine when the calling party line is available to receive a call connection. - Since many STN subscribers have only single line access to the STN, they use the single line to connect to both the
Internet 34 for data services and theSTN 10 for voice services. Consequently, when a call request is received by theVSP 48, theVSP 48 may automatically send a TCAP query to the calling party's SP. Alternatively, the form completed by the callingparty using PC 32 may include an indication of whether the calling party has single line access to the STN. In that instance, the TCAP query shown in Fig. 2 is sent only when the form indicates that the calling party has single line access to the STN. If the calling party has multiple line access to the STN, a TCAP query is not sent unless a first call attempt to the calling party'stelephone 30 is unsuccessful. - The TCAP query message sent to the
SSP 18 is a Query with Permission (QUERY + P). The query message is sent to request that the calling party's SSP monitor the calling party's telephone line to determine when the line becomes idle. Normally, theSSP 18 will respond with an acknowledgement message in the form of a TCAP message type known as Conversation with Permission (CONVERSATION + P) to indicate to the VSP that the originating switch will perform scanning of the calling line. When the calling line becomes available,SSP 18 will return a TCAF RESPONSE message to theVSP 48 to inform theVSP 48 that the calling party's line is available. As an alternative to this sequence, theVSP 48 may use a QUERY + P message to query the busy/idle status of the calling party's line. That query normally returns an immediate TCAP RESPONSE from theSSP 18 with an indication of the busy/idle status of the calling party line. The busy/idle status query may be used as an initial inquiry, rather than a request for scanning of the line. The choice of initial queries is dependent on the information available to the VSP, as well as the elapsed time since receipt of the call request message. - In either case, when the
VSP 48 has received confirmation that the calling party line is available, theVSP 48 sends a first IAM with a circuit identification code (CIC) = "001", for example as described in applicant's co-pending application incorporated herein by reference. On receipt of the IAM, theSSP 18 checks the availability of the calling party's line and returns an Address Complete (ACM) message with a CIC=001 to theVSP 48 indicating that the calling party line is available. Concurrently, theSSP 18 applies a ringing signal to the calling party line which causes the calling party'stelephone 30 to ring. - When the calling party answers the
telephone 30, theSSP 18 returns an Answer (ANM) message to theVSP 48 with the CIC=001. On receipt of the ANM message, theVSP 48 immediately formulates and transmits an IAM message with a CIC=002 (the opposite end of the loop-back trunk 64) to theSSP 18. TheSSP 18 receives that IAM and assumes that there is an incoming call on the loop-back trunk 64. It examines the called number in the IAM and determines that the caller is served by another SSP in theSTN 10,SSP 14 for example. TheSSP 18 therefore formulates an appropriate IAM and transmits it into the signaling network of theSTN 10.STN 10 forwards the IAM to theSSP 14 which checks the availability of the called party line. On finding the called party lineavailable SSP 14 returns an ACM message to theSSP 18. Concurrently, theSSP 14 applies a ringing signal totelephone 26. On receipt of the ACM message atSSP 18, theSSP 18 returns to theVSP 48 an ACM message with a CIC=002. - This completes the call connection between the calling party's
telephone 30 and the called party'stelephone 26. When the called party attelephone 26 answers the call, an ANM is returned through theSTN 10 to theSSP 18 which in turn returns the ANM to theVSP 48. Since the call connection is completed, conversation between the calling and the called party attelephones - In this example, a disconnect is received from the calling
party 30. When the callingparty 30 goes on-hook, theSSP 18 sends a release message with a CIC=001 to theVSP 48. Since theVSP 48 recognizes that the release message belongs to a call which it initiated, the VSP discards the release message and returns a release complete (RLC) with CIC=001 to theSSP 18. The VSP likewise formulates and returns an REL message with a CIC=002 to theSSP 18 in order to release the second half of the call. On receipt of the REL message,SSP 18 returns an RLC message with CIC=002 to theVSP 48 and formulates and sends an REL message through theSTN 10 to theSSP 14. On receipt of the REL message, theSSP 14 returns an RLC message to theSSP 18. Thereafter it applies dial tone to the telephone line ofuser 26 which responds by going on-hook and the call series is completed. - As described in applicant's co-pending application, the call request initiated by the calling
party using PC 32 may have included several called numbers in which case the calling party would normally not be disconnected from theSSP 18 and the VSP on receipt of a release from the called party's SSP would proceed with setting up a call connection to the next called number. - Figs. 3a and 3b show a portion of a call message flow involved in the handling of a call request initiated by a calling party at
PC 32. As described above, the callingparty using PC 32 accessesweb server 58 and initiates a call request which is sent through the Internet to theweb server 58 and forwarded through the Internet to theVSP 48. In this example, the call request indicates that the callingparty using PC 32 has single line access to theSTN 10. TheVSP 48 therefore forwards a TCAP QUERY + P message to theSSP 18 requesting that theSSP 18 scan the calling party's telephone line to determine when the line becomes available to receive a call. TheSSP 18 responds with an acknowledgement message, as described above, followed when the calling line becomes available with a TCAP RESPONSE message indicating that the calling line is available to receive a call. As also described above, this prompts theVSP 48 to initiate a call setup in which theVSP 48 connects a first leg of the call connection with the calling party attelephone 30 and then initiates a second leg of the call connection by sending an IAM through theSTN 10 to theSSP 14. - In this example, the called
line 26 is busy and theSSP 14 returns a release message to theSSP 18. On receipt of the REL message from theSSP 18, theSSP 18 formulates an REL message with a CIC=002 and forwards it to theVSP 48. TheSSP 14 also returns an RLC message to theSSP 14 through theSTN 10. On receipt of the REL message, theVSP 48 returns an RLC to theSSP 18 with a CIC=002.VSP 18, having received information that the called party attelephone 26 is not available, and that the calling party attelephone 30 has a single line connection to theSTN 10, is programmed to connect thetelephone 30 to an IVR 62 (Fig. 1) in order to obtain information from the calling party about how to proceed. To accomplish this, the VSP sends an IAM with a CIC=002 to theSSP 18 and a called number of theIVR 62. TheSSP 18 determines that theIVR 62 is served by another SSP in theSTN 10. It therefore forwards an IAM through theSTN 10 to theSSP 14 which serves theIVR 62. Ensuing ACM and ANM messages are exchanged, as illustrated. Those messages follow the same sequence as the ACM and ANM described with reference to the top half of the diagram. After sending the IAM to connect the calledparty telephone 30 to theIVR 62, the VSP also sends a data message to theIVR 62 informing theIVR 62 of the call and requesting that theIVR 62 begin an interactive session with the callingparty telephone 30 in order to get instructions as to how the balance of the call request session is to be handled. When the call connection with theIVR 62 is effected, the IVR plays a pre-recorded message to the calling party attelephone 30 requesting that the calling party select an option regarding how the calling session should proceed. A few of the possible options which may be enabled are as follows: - a) the call attempt to the called party may be abandoned and the next called party number in the call request message attempted;
- b) the calling party line may be held open and a
request to monitor the called party line sent to the
VSP 48, in which case the calling party is held at the IVR until the called party line becomes available; - c) the calling party may request that they be
disconnected from the call, but the called party
line monitored, in which case the
VSP 48 would attempt to recomplete the call connection once the called party line becomes available; - d) the call may be abandoned but the called party line monitored, if the calling party has a call waiting feature. In that case, as soon as the called party line became available the VSP completes a call connection between the calling and called party lines, and the calling party uses the call waiting feature to connect to the called party line;
- e) the calling party may specify a time interval during which VSP should monitor the called party line and if the called party line becomes available during the time interval, the VSP attempts to complete the call connection between the calling and the called party lines.
-
- It should be understood by those skilled in the art that the five options described above are exemplary only. Other options may be used or only some of the options described above need be implemented.
- If the calling party selects option a), the call is simply abandoned and control is returned to the VSP which attempts to complete the call connection between the calling party and the next called party umber in the call request message.
- If the calling party selects option b), a TCAP query is sent by the
VSP 48 to the SSP which serves the calling party requesting that the SSP scan the called party line and advise the VSP as soon as the called party line becomes available to accept the call. In the meantime, the call is held by the IVR. TheIVR 62 preferably plays auditory content to the caller while the caller is held. The auditory content may be any desirable content including recorded music, a radio station, advertisements, or interactive entertainment which theIVR 62 is programmed to provide. - If the user selects option c), the user is aware that there are no further called numbers in the call request message and the user prefers not to be held on the line while waiting for the called party to become available. The
VSP 48 therefore disconnects the calling party line from theIVR 62 and sends a query message to theSSP 14 to monitor the called party line. When theSSP 14 returns a RESPONSE message indicating that the called party line is available, the VSP attempts to create a call connection between the calling party line and the called party line as described above with reference to Fig. 2. - If the user elects option d), the call attempt is abandoned and the VSP attempts to establish a call connection between the calling party and the next called party in the call request message. The VSP simultaneously sends a TCAP QUERY + P message to the
SSP 14 through theSTN 10 requesting that theSSP 14 monitor the called party line and notify theVSP 48 when the called party line becomes available to receive a call. When theVSP 48 receives a RESPONSE message from theSSP 14 indicating that the called party line is available, theVSP 48 preferably formulates a first IAM to establish a connection with the callingparty telephone 30 followed immediately by a second IAM to establish a connection with the calledparty telephone 26, without waiting for the receipt of ACM and ANM messages. Since the calling party is known to have a call waiting feature, it is assumed that the calling party will respond to the call. Since theVSP 48 has just been notified that the called party line is available, there is an excellent probability that the called party will answer the call. - If the user selects option e), the user must specify a time interval when the
VSP 48 should monitor the called party line for availability. This option would normally be used by a calling party which expects to be available during a certain period of the day, say from 2:00-4:00 p.m. On receipt of option e) returned from theIVR 62, theVSP 48 queues a monitoring request. When its clock indicates 2:00 p.m., theVSP 48 sends a TCAP QUERY + P message to theSSP 14 requesting that it report the line status of the called party. If theSSP 14 reports that the called party line is busy, theVSP 48 returns a TCAP QUERY + P message requesting that theSSP 14 scan the called party line and advise theVSP 48 when the line becomes available. If theSSP 14 responds before 4:00 p.m. that the calling party line is available, theVSP 48 responds by formulating a first IAM which initiates a call request to the calling party line. After the calling party answers, theVSP 48 initiates a second IAM to complete a second leg of the call connection as described with reference to Fig. 1. - Returning again to Fig. 3a, after the
IVR 62 plays the called party busy options to the calling party attelephone 30, the calling party returns DTMF signals by keying a selection which indicates option b) because the callingparty 30 urgently wishes to reach the called party attelephone 26. Those instructions are forwarded via a data connection from theIVR 62 to theVSP 48. On receipt of the instructions, theVSP 48 immediately launches a TCAP QUERY + P message to theSSP 14 through theSTN 10 requesting that theSSP 14 monitor the line oftelephone 26. TheSSP 14 returns an acknowledgement through theSTN 10 and begins a scan of the called party line. Whentelephone 26 goes on-hook, theSSP 14 returns a TCAP RESPONSE message through theSTN 10 advising theVSP 48 that the calling party line is available. - As shown in Fig. 3b, immediately on receipt of the TCAP RESPONSE, the
VSP 48 formulates an IAM with a CIC=002 and a called number of thetelephone 26 which it forwards toSSP 18. On receipt of the IAM, theSSP 18 formulates an IAM which it forwards through theSTN 10 to theSSP 14. On receipt of the IAM, theSSP 14 checks the availability of the calledparty line 26 and, finding it available, returns an ACM message to theSSP 18 which causes theSSP 18 to formulate an ACM message with a CIC=002 which it returns to theVSP 48. Concurrently, theSSP 14 applies a ringing signal to the calling party line which causes thetelephone 26 to ring. When the calling party answers, theSSP 14 returns the ANM through theSTN 10 to theSSP 18 which in turn forwards an ANM with a CIC=002 toVSP 48. This establishes a connection between the callingparty telephone 30 and the calledparty telephone 26 and conversation ensues. Thereafter, the call progresses in accordance with the remainder of the call request message which may initiate further calls between the parties or may release the connections as described above with reference to Fig. 2. - Fig. 4 shows an exemplary call flow sequence in which VSP 46 (see Fig. 1) is used as an out-dialer to set up calls for a
call center 70. Because of the flexibility offered by the VSP,VSP 46 can perform all call setup functions for thecall center 70 and because of its position in the signaling network, theVSP 46 is enabled to use PSTN resources much more efficiently than most call center equipment in use today. - In the exemplary call flow sequence shown in Fig. 4,
call center 70 sends a called number list to theVSP 46 via a data connection, for example the data communications link 74 (see Fig. 1). The called number list may be supplied at any time and may augment a called number queue already resident onVSP 46. In this example, a called number on the called number list istelephone 66 served by SSP 18 (see Fig. 1). TheVSP 46 therefore formulates an IAM with a CIC=001 and a called number set to the number of thetelephone 66 and forwards the IAM through theSTN 10 to theSSP 18. In response, theSSP 18 checks the availability oftelephone 66 and determines that it is busy. TheSSP 18 therefore returns an ISUP Release (REL) message indicating to theVSP 46 that the call cannot be completed.VSP 46 responds with a Release Complete (RLC) message indicating that CIC=001 has been released. - In order to determine the status of
telephone 66, the VSP sends a TCAP Query with Permission (Q+P) message to theSSP 18 requesting that theSSP 18 monitor the called party line and advise when the line becomes idle.SSP 18 responds to the TCAP message with a TCAP Conversation with Permission (C+P) shown in the diagram as an "ACK" response. - When the called
party telephone 66 goes on-hook (not illustrated), theSSP 18 sends a TCAP response message to theVSP 46 reporting that the called party line is idle.VSP 46 responds to the message by sending a data query message to thecall center 70 to determine whether an agent is available to take the call. Alternatively, the VSP may maintain an agent status file in which it maintains the current status of agents that is routinely updated by the call center, or theVSP 46. In response to the query message, thecall center 70 responds that there is an agent available to take the call. Consequently, theVSP 46 launches an IAM with a Circuit Identification Code (CIC) of "001" to connect with the called party. TheSSP 18 verifies that the called party line fortelephone 66 is available and returns an ACM with the CIC=001. Concurrently, theSSP 18 applies a ringing signal to the called party line. When the called party answers, theSSP 18 returns an ANM message with the CIC=001. On receipt of the ANM message, theVSP 46 sends a second IAM with a CIC=002 and the called number a number for thecall center 70. When theSSP 18 receives the IAM it assumes that it is associated with an incoming call arriving on the loop-back trunk 64 and examines the called number to determine how the call should be routed. The called number indicates that the call should be routed to theSSP 12 which serves thecall center 70. TheSSP 18 therefore formulates an IAM which is forwarded through theSTN 10 to theSSP 12.SSP 12 forwards the IAM to thecall center 70 which in this example is assumed to be served by an ISDN trunk with PRI signaling. The ACD or PBX at the call center returns an ACM indicating that an agent phone is available to take the call and an ACM message is sent bySSP 12 back to theSSP 18 which returns an ACM to theVSP 46. Concurrently, the call center applies a ringing tone or some other advisement message to the agent attelephone 76. When the agent takestelephone 76 off-hook, the ACD or PBX at the call center returns an ANM message to theSSP 12 which sends an ANM message to theSSP 18. TheSSP 18 sends an ANM message to theVSP 46 with a CIC=002, which permits the VSP to deduce that the call connection is complete. Thereafter, the called party attelephone 66 is connected with the agent attelephone 76 and conversation ensues. - When conversation is completed, one of the parties will go on-hook which will cause a release sequence similar to that shown in the lower half of Fig. 3b.
- It should be understood that there is no requirement to release agents between calls. The agent's connection through
SSP 12 to the CIC=002 of loop-back trunk 64 may therefore be maintained for an extended time to further conserve signaling resources. If the agents connection is maintained between calls, the data communications link 74 may be used by an agent to indicate to theVSP 46 that a called party should be disconnected. If theVSP 46 receives an indication to that effect, theVSP 46 can send an ISUP REL message with a CIC=001 and the calling and called party numbers to effect the release. Thus agent efficiency can be further improved, while call control is maintained. - Fig. 5 further illustrates the use of a
VSP 46 as an out-dialer forcall center 70. In particular, it illustrates efficient use of the STN enabled by theVSP 46 when configured to function as an out-dialer for thecall center 70. - In the first example shown in Fig. 5, the
VSP 46 receives a call list from thecall center 70 and commences a call from the list by formulating a first IAM with a CIC=001 that it forwards to theSSP 18. The called number is to a number served by theSSP 18 which is no longer in service. TheSSP 18 responds by returning an ISUP REL message with a cause indicating that the number is unavailable. TheVSP 46 is enabled to interpret ISUP Release Causes and recognizes the cause=Unavailable as an indication that the number is no longer in service. TheVSP 46 therefore formulates a RLC message with CIC=001 and returns the RLC message to theSSP 18. Concurrently, theVSP 46 sends a data message over the data connection 74 (see Fig. 1) to thecall center 70 indicating that that called number is no longer available and it should be deleted from all called number lists. Alternatively, theVSP 46 may log such numbers and include them in an exception report which is periodically sent over the data communications link 74 to thecall center 70 or reported in some other manner. - In a second example, an IAM including a new called number is formulated by the
VSP 46. The IAM has a CIC=001 and includes a called number from the call list. On receipt of the IAM, theSSP 18 returns an ACM with a CIC=001 indicating that the called number is available. Concurrently, theSSP 18 applies ringing of the called party line. In this example, the called party line has a call waiting feature and a voice mail service. When the call is not answered, the call is forwarded to the voice mail service. This prompts theSSP 18 to return a Pass Along Message (PAM) with a CIC=001 to theVSP 46. Parameters in the PAM indicate to theVSP 46 that the call has been forwarded to another number. TheVSP 46 therefore responds by sending a Release message with CIC=001 to theSSP 18 which terminates ringing to the forwarded number and SSP 18returns an RLC message indicating that the call has been cancelled and release is complete. Since theVSP 46 cannot determine why the call was forwarded from the contents of the PAM message, the VSP forwards a TCAP Q+P message requesting the status of the called party line. TheSSP 18 responds with TCAP Response message that the called party line is busy. Consequently, the VSP formulates a TCAP Q+P message to theSSP 18 requesting that theSSP 18 monitor the called party line and report when the called party line becomes idle. Eventually, theSSP 18 returns a TCAP response indicating that the called party line is idle. - On receipt of the indication by the
SSP 18 that the called party line is idle,VSP 46 may query thecall center 70, or a list theVSP 46 maintains, to determine if an agent is available to take the call. TheVSP 46 completes the call to the called party if the agent is available, as explained above with reference to Fig. 4. Alternatively, theVSP 46 may add the call to the head of the memory queue of numbers to be called. If, however, the TCAP response received by theSSP 18 indicates that the called party line is idle rather than busy, theVSP 46 is enabled to deduce that the called party line is unattended. The called number may therefore be placed in a special queue to be called at a later time. Algorithms may be written to determine when calls placed in a special queue are to be retried. - As an alternative, the call may be placed at the bottom of the memory queue of numbers to be called and retried when it arrives again at ahead of the queue. As a further alternative, a call attempt count associated with the number may be incremented when the number is placed at the bottom of the queue. If this option is used, every time a call attempt is made using a call from the memory queue of numbers to be called, the call attempt count is examined to determine whether it has exceeded a predetermined limit, and the called number is removed from the list and added to an exception report, or the like, if the call attempt count exceeds some predetermined value.
- The
VSP 46 therefore provides an excellent out-dialer for a call center which permits calls to be placed much more economically and intelligently. It also provides an economical mechanism for removing unavailable numbers from called number lists and for efficiently and accurately handling calls to numbers where no called party is available. - The system and the method in accordance with the invention therefore provide additional functionality in the STN which permits service providers to offer more options to calling parties wishing to automate call completion. It also minimizes common channel signaling by ensuring that call setup messages are sent at opportune times to ensure with reasonable certainty that call completion will ensue.
- Changes and modifications to the above-described embodiments will no doubt become apparent to persons skilled in the art. The scope of the invention is therefore intended to be limited solely by the scope of the appended claims.
Claims (35)
- A method of completing a voice connection between a first and second voice terminal on a Switched Telephone Network (STN) (10), including the steps of receiving at a Virtual Switching Point (VSP) (48) a call request message from a calling party (30, 32), the call request message being received through a connection to a data network (34); sending a first Common Channel Signaling (CCS) message from the VSP over a signaling network of the STN to an SSP (18) in the STN to initiate a first leg of the voice connection, and sending a second CCS message from the VSP to an other SSP in the STN to initiate a second leg of the voice connection, the first and second CCS messages being interrelated to an extent that a circuit identification code in each message is associated with opposite ends of the same trunk facility, CHARACTERIZED by:sending a Transactions Capability Application Part (TCAP) query message from the VSP (48) to the SSP (18) that serves the calling party to determine the availability of a calling line identified in the call request message, and initiating a call from the VSP using Integrated Services Digital Network User Part (ISUP) common channel signaling messages sent from the VSP to initiate the first and second legs of the voice connection, if a response to the TCAP query message indicates that the calling line is available.
- A method as claimed in claim 1 wherein the calling party has a single line connection to the STN, the single line connection being used to send the call request message over the data network to the VSP (48), and the query message is used to determine when the calling party disconnects from the data network (34) after sending the call request message, in order to minimize a number of common channel signaling messages required to set up a call to the calling party number.
- A method as claimed in claim 2 wherein the TCAP query message is a query with permission (QUERY + P) message sent from the VSP (48) to the SSP (18) to determine the availability of the calling party line.
- A method as claimed in claim 3 wherein the SSP (18) responds to the QUERY + P message with a conversation with permission (CONVERSATION + P) TCAP message to acknowledge the query message and indicate to the VSP (48) that the SSP will scan the calling party to determine when the calling party line is available for call setup.
- A method as claimed in claim 4 wherein when the calling party line becomes available, the SSP (18) responds the to the QUERY + P message using a TCAP RESPONSE message to advise the VSP (48) that the calling party line is available for call setup.
- A method as claimed in any preceding claim wherein initiating the call connection between the calling and called party numbers involves:sending a first ISUP Initial Address Message (IAM) from the VSP (48) to initiate the first leg of the call connection, the called number in the first IAM being associated with the calling party line;receiving from the SSP (18) that serves the calling party number an ISUP Address Complete (ACM) and Answer (ANM) messages indicating that the calling party has answered and the first leg of the call connection is complete;sending a second ISUP IAM from the VSP to initiate the second leg of the call connection, the called number in the second IAM being the called party number; andreceiving from an SSP (14) that serves the called party number an ISUP ACM and an ANM message to indicate that the called party has answered the second leg of the call connection and the call connection is complete.
- A method as claimed in any preceding claim wherein if the call connection initiated between the calling and the called party numbers is not completed because a called party line associated with the called party number is busy, the VSP (48) initiates a request for instructions from the calling party to determine further treatment of the call.
- A method as claimed in claim 7 wherein the VSP (48) initiates a request for instructions by sending an ISUP IAM from the VSP to connect the calling party with an Interactive Voice Response (IVR) unit (62) in the STN (10) and the IVR prompts the calling party to select one of a number of predefined options respecting farther treatment of the call.
- A method as claimed in claim 8 wherein the predefined options include:a) abandon the call and attempt the next called party number in the call request message;b) hold the calling party line open and monitor the called party line until the called party line becomes available;c) disconnect the calling party line, monitor the called party line and re-attempt the call connection when the called party line becomes available;d) abandon the call and attempt a next call in the call request message, but monitor the called party line and call the calling party line as soon as the called party line becomes available because the calling party has a call waiting feature;e) abandon the call and attempt the call connection again if the called party line is available at any time during a specified time period.
- A method as claimed in claim 9 wherein if option b) is selected, the VSP (48) sends a TCAP QUERY + P message to the SSP (18) that serves the called party, the connection with the IVR (62) is maintained until the called party line becomes available or the calling party terminates the connection, and the IVR plays audible content to the calling party on hold.
- A method as claimed in claim 9 wherein if option c) is selected the VSP (48) sends a TCAP QUERY + P message to the SSP (18) which serves the called party after the calling party line is disconnected, and on receipt of a RESPONSE message indicating that the called party line is available, the VSP sends a first ISUP IAM in which the called number is the calling party number to initiate the first leg of the call connection and on receipt of an ACM and ANM messages in response to the IAM, the VSP sends a second IAM in which the called number is the called party number in order to complete the second leg of the call connection.
- A method as claimed in claim 9 wherein if the option d) is selected, the VSP (48) sends a TCAP QUERY + P message to the SSP (18) which serves the called party after the IVR (62) is disconnected, and on receipt of a RESPONSE message indicating that the called party line is available, the VSP sends a first ISUP IAM in which the called number is the calling party number to initiate the first leg of the call connection and a second IAM in which the called number is the called party number in order to complete the second leg of the call connection without waiting for an ACM or an ANM message from either end of the call connection.
- A method as claimed in claim 9 wherein if the option e) is selected the VSP (48) sends a TCAP QUERY + P message to the SSP (18) that serves the called party at a start of the specified time period, and if a TCAP RESPONSE message indicating that the called party line is available before an end of the specified time period, the VSP sends a first ISUP IAM in which the called number is the calling party number to initiate the first leg of the call connection and on receipt of an ACM and an ANM messages to indicate that the called party has answered, the VSP sends a second ISUP IAM in which the called number is the called party number in order to complete the second leg of the call connection; and
if the calling party does not answer in response to the first ISUP IAM or the specified time expires before the calling party line becomes available, the VSP deletes the request and frees associated resources. - A system for using TCAP signaling for improved call set up in a switched telephone network (STN) (10), that includes a virtual switching point (VSP) (48) which is a physical node in a common channel signaling network of the STN and a logical node in a transport network of the STN, a plurality of service switching points (SSPs) (12-18) in the STN having TCAP signaling capability, and a trunk facility (20, 64) in the STN, the VSP being a logical switching node logically located between physical ends of the trunk facility, CHARACTERIZED by:a server (58) connected to a data network (34) and accessible through the data network, the server accepting call requests from calling parties having access to the data network;the VSP (48) is enabled with Transactions Capability Application Part (TCAP) signaling capability and the VSP has a connection to the data network and receives the call requests from the server, the VSP setting up call connections between the calling party and a called party specified in the call request to a called party number by sending a first Integrated Services Digital Network User Part (ISUP) signaling message to the SSP (18) serving the calling party, and another ISUP signaling message to the SSP (12) serving the called party so that the call setup between the called party and the calling party is routed over the trunk facility; anda gateway switch located between the VSP and the data network to isolate the VSP from the data network to prevent direct access to the VSP form the data network.
- A system as claimed in any one of claims 15 wherein the system further includes an interactive voice response unit (IVR) (62) in the STN (10), the IVR accepting calls set up by the VSP (48) between the calling party and the IVR when a call connection cannot be set up by the VSP between the calling party and a called party specified in a call request message.
- A system as claimed in claim 15 wherein the IVR (62) has a data link with the VSP (48) and the IVR is enabled to communicate an option respecting further call handling which is chosen by the calling party in response to prompting messages presenting a number of predefined options for further call handling.
- A system as claimed in claim 16 wherein a one of the predefined options permits the calling party to stay on hold while the called party line is monitored for availability to take a call from the calling party and the IVR (62) is enabled to play audible content during a period that the calling party is held by the IVR.
- A method of completing calls through a switched telephone network for a call center (70) having a predetermined list of numbers to be called, including the steps of placing the list of numbers to be called in a memory queue which may be accessed by a virtual switching point (VSP) (48) in the network, formulating a first Integrated Services Digital Network User Part (ISUP) signaling Initial Address Message (IAM) at the VSP, the IAM containing a called number from the queue, and forwarding the IAM to initiate a call connection with the called number, CHARACTERIZED by:on receipt of an ISUP Address Complete (ACM) message and an ISUP Answer (ANM) message in response to the first IAM, formulating a second IAM at the VSP (48), the second IAM including a number of the call center (70) as the called number and forwarding the second IAM to connect the called number from the queue with the call center, the first and second ISUP IAM messages being related to an extent that the call is routed ever a trunk facility in the switched telephone network in which the VSP is a logical switching mode located between opposite ends of the trunk facility.
- The method as claimed in claim 18 wherein, if an ISUP Release (REL) message is received at the VSP (48) in response to the first IAM, the Cause associated with the Release message is examined to determine a course of action respecting a further treatment of the called number.
- The method as claimed in claim 19 wherein if the ISUP REL message indicates that a line associated with the called number is busy, the VSP (48) sends a TCAP message to a switching point (12-18) that serves the line requesting that the line be monitored to determine when the line becomes idle.
- The method as claimed in claim 20 wherein on receipt of a TCAP response message indicating that the line is idle, the VSP (48) checks the availability of an agent at the call center (70) to receive a call and the VSP sends a first IAM if an agent is available to recsive the call.
- The method as claimed in claim 21 wherein the VSP (48) checks the availability of an agent to take the call by sending a data query message to an automatic call distributor or a private branch exchange at the call center (70).
- The method as claimed in claim 22 wherein a predictive algorithm is used to determine the availability of an agent to receive the call.
- The method as claimed in any one of claims 21-23 wherein on receipt of a TCAP response message indicating that the line is idle, the VSP (48) places the called number associated with the response at the head of the memory queue of numbers to be called.
- The method as claimed in any one of claims 19-24 wherein if the cause associated with the ISUP REL message indicates that the number is unavailable, the number is removed from the list of numbers to be called and a data message is sent to the call center (70) indicating that the number is unavailable.
- The method as claimed in claim 28 wherein the message sent to the call center (70) is a part of an exception report indicating all numbers removed from the list of numbers to be called and a reason that each number was removed from the list.
- The method as claimed in any one of claims 19-26 wherein if an ISUP Pass Along Message (PAM) is received in response to the first IAM indicating that the call has been forwarded to another number, an ISUP REL message is sent back to a switching point (12-18) that serves the called number to cancel the call.
- The method as claimed in claim 27 wherein after the ISUP REL message is sent, a TCAP message is sent by the VSP (48) to the switching point (12-18) to determine a status of a line associated with the called number.
- The method as claimed in claim 28 wherein if a response to the TCAP message indicates that the line associated with the called number is busy, a TCAP message is sent to the SSP (12-18) requesting that the line be monitored to determine when the line becomes idle.
- The method as claimed in claim 29 wherein on receipt of a TCAP response message indicating that the line is idle, the VSP (48) checks the availability of an agent at the call center (70) to receive a call and the VSP sends a first IAM if an agent is available to receive the call.
- The method as claimed in claim 28 wherein if the response to the TCAP message indicates that the line is idle, the called number is placed at a bottom of the memory queue.
- The method of completing calls to a call center as claimed in claim 31 wherein if the called number is placed at a bottom of the memory queue, a call attempt count associated with the number is incremented and the called number is removed from the memory queue and added to an exception report when the call attempt count reaches a predetermined value.
- An out-dialer for a cell center (70), CHARACTERIZED by:a virtual switching point (VSP) (46) which is a physical node in a signaling network of a switched telephone network (STN) (10) that serves the call center (70), and a virtual node in at least one trunk facility of the STN;a memory queue accessible by the VSP, the memory queue containing at least one list of called numbers to which calls are to be set up by the VSP; andthe VSP (46) sets up calls to called numbers by sending a first ISUP IAM message to a switching point associated with an end of the trunk facility in which the VSP is the virtual node, the first ISUP IAM message containing the called number; andthe VSP (46) completes the set up of the call by sending a second ISUP IAM message to a switching point associated with an opposite end of the trunk facility in which the VSP is the virtual node, the second IAM including a called number that is a number of the call center (70).
- An out-dialer as claimed in claim 33 wherein the VSP (46) has a data communications link with the call center (70) to permit data messages to be exchanged between the VSP and the call center.
- An out-dialer as claimed in claim 33 wherein after a initial connection with an agent at a call center (70), the VSP (46) does not disconnect the agent between calls unless explicitly instructed to do so by a data message sent over a data coininanications link (74) with the call center.
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
US71092 | 1998-05-01 | ||
US09/071,092 US6236722B1 (en) | 1998-05-01 | 1998-05-01 | Method and system for using TCAP signaling for improved call setup from a virtual switching point |
PCT/CA1999/000398 WO1999057915A1 (en) | 1998-05-01 | 1999-05-03 | Method and system for improved call setup |
Publications (2)
Publication Number | Publication Date |
---|---|
EP1075768A1 EP1075768A1 (en) | 2001-02-14 |
EP1075768B1 true EP1075768B1 (en) | 2003-01-15 |
Family
ID=22099201
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP99919006A Expired - Lifetime EP1075768B1 (en) | 1998-05-01 | 1999-05-03 | Method and system for improved call setup |
Country Status (9)
Country | Link |
---|---|
US (1) | US6236722B1 (en) |
EP (1) | EP1075768B1 (en) |
AT (1) | ATE231314T1 (en) |
AU (1) | AU3696199A (en) |
BR (1) | BR9910170A (en) |
CA (1) | CA2270601C (en) |
DE (1) | DE69904931T2 (en) |
ES (1) | ES2192381T3 (en) |
WO (1) | WO1999057915A1 (en) |
Families Citing this family (35)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US6944184B1 (en) * | 1998-12-04 | 2005-09-13 | Tekelec | Methods and systems for providing database node access control functionality in a communications network routing node |
US7050456B1 (en) * | 1998-12-04 | 2006-05-23 | Tekelec | Methods and systems for communicating signaling system 7 (SS7) user part messages among SS7 signaling points (SPs) and internet protocol (IP) nodes using signal transfer points (STPs) |
US6324183B1 (en) | 1998-12-04 | 2001-11-27 | Tekelec | Systems and methods for communicating messages among signaling system 7 (SS7) signaling points (SPs) and internet protocol (IP) nodes using signal transfer points (STPS) |
US7002988B1 (en) * | 1998-12-04 | 2006-02-21 | Tekelec | Methods and systems for communicating SS7 messages over packet-based network using transport adapter layer interface |
US6690664B1 (en) * | 1999-04-27 | 2004-02-10 | Sprint Communications Company, L.P. | Call center communications system for handling calls to a call center |
US7536002B1 (en) | 1999-07-09 | 2009-05-19 | Jpmorgan Chase Bank, National Association | System and method of intelligent call routing for cross sell offer selection based on optimization parameters or account-level data |
EP1131943A1 (en) * | 1999-09-24 | 2001-09-12 | Nokia Corporation | Ip telephony system and method of operation thereof using ss7 network |
US6782417B1 (en) * | 1999-10-12 | 2004-08-24 | Nortel Networks Limited | Advertising system for callers to busy data service providers |
US7318091B2 (en) * | 2000-06-01 | 2008-01-08 | Tekelec | Methods and systems for providing converged network management functionality in a gateway routing node to communicate operating status information associated with a signaling system 7 (SS7) node to a data network node |
CN100347987C (en) * | 2000-09-22 | 2007-11-07 | 桑特拉系统有限公司 | System and method for distributed multi-party call control |
US6647108B1 (en) * | 2001-01-02 | 2003-11-11 | Verizon Services Corp. | Internet call manager |
US20040024892A1 (en) * | 2001-05-21 | 2004-02-05 | Creswell Carroll W. | System for providing sequenced communications within a group |
US7257620B2 (en) * | 2001-09-24 | 2007-08-14 | Siemens Energy & Automation, Inc. | Method for providing engineering tool services |
US6959081B2 (en) * | 2001-12-03 | 2005-10-25 | International Business Machines Corporation | Expert hold queue management |
US7139390B2 (en) * | 2001-12-12 | 2006-11-21 | International Business Machines Corporation | Promoting use of experts to callers waiting in a hold queue |
US6956935B2 (en) * | 2001-12-17 | 2005-10-18 | International Business Machines Corporation | Origin device billing according to caller |
US6977998B2 (en) | 2001-12-17 | 2005-12-20 | International Business Machines Corporation | Destination device billing according to call recipient |
US6834103B1 (en) * | 2002-03-29 | 2004-12-21 | Bellsouth Intellectual Property Corporation | Caller control of call waiting services |
US7158630B2 (en) * | 2002-06-18 | 2007-01-02 | Gryphon Networks, Corp. | Do-not-call compliance management for predictive dialer call centers |
US7085821B2 (en) * | 2002-06-14 | 2006-08-01 | International Business Machines Corporation | TCAP event processing environment |
US7822609B2 (en) * | 2002-06-14 | 2010-10-26 | Nuance Communications, Inc. | Voice browser with integrated TCAP and ISUP interfaces |
US7804789B2 (en) | 2004-03-18 | 2010-09-28 | Tekelec | Methods, systems, and computer program products for organizing, managing, and selectively distributing routing information in a signaling message routing node |
US8126017B1 (en) * | 2004-05-21 | 2012-02-28 | At&T Intellectual Property Ii, L.P. | Method for address translation in telecommunication features |
US7532647B2 (en) * | 2004-07-14 | 2009-05-12 | Tekelec | Methods and systems for auto-correlating message transfer part (MTP) priority and internet protocol (IP) type of service in converged networks |
US20060083367A1 (en) * | 2004-10-19 | 2006-04-20 | Schepers Paul D | Transaction capabilities application part message router |
WO2006056256A2 (en) * | 2004-11-19 | 2006-06-01 | Richard Bergner Verbindungstechnik Gmbh & Co Kg | Hydraulic unit and method for providing a pressurized hydraulic fluid |
US8175939B2 (en) * | 2005-10-28 | 2012-05-08 | Microsoft Corporation | Merchant powered click-to-call method |
US8411833B2 (en) * | 2006-10-03 | 2013-04-02 | Microsoft Corporation | Call abuse prevention for pay-per-call services |
US20080159521A1 (en) * | 2006-12-29 | 2008-07-03 | Dave Sneyders | System For Establishing Outbound Communications With Contacts From A Call Center |
WO2009011563A1 (en) * | 2007-07-13 | 2009-01-22 | Telefonaktiebolaget Lm Ericsson (Publ) | Method of routing a call made to a fixed telephone number of a uma-subscriber in a telecommunication network |
US9043451B2 (en) * | 2007-07-31 | 2015-05-26 | Tekelec, Inc. | Methods, systems, and computer readable media for managing the flow of signaling traffic entering a signaling system 7 (SS7) based network |
US9219677B2 (en) | 2009-01-16 | 2015-12-22 | Tekelec Global, Inc. | Methods, systems, and computer readable media for centralized routing and call instance code management for bearer independent call control (BICC) signaling messages |
US8527598B2 (en) * | 2010-02-12 | 2013-09-03 | Tekelec, Inc. | Methods, systems, and computer readable media for answer-based routing of diameter request messages |
CN109039998A (en) * | 2017-06-12 | 2018-12-18 | 中兴通讯股份有限公司 | Method of calling and device |
US11397198B2 (en) | 2019-08-23 | 2022-07-26 | Schweitzer Engineering Laboratories, Inc. | Wireless current sensor |
Family Cites Families (13)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US5282244A (en) | 1991-06-24 | 1994-01-25 | At&T Bell Laboratories | Virtual signaling network method |
US5377186A (en) * | 1993-07-21 | 1994-12-27 | Telefonaktiebolaget L M Ericsson | System for providing enhanced subscriber services using ISUP call-setup protocol |
US5479495A (en) * | 1993-10-01 | 1995-12-26 | U S West Advanced Technologies, Inc. | Method and system for automatically accessing and invoking switch-based services in an advanced intelligent network |
US5425091A (en) * | 1994-02-28 | 1995-06-13 | U S West Technologies, Inc. | Method and system for providing an automatic customer callback service |
US5600710A (en) * | 1994-07-08 | 1997-02-04 | Bellsouth Corporation | Method for providing a recorded message to a telephone caller when called number is busy |
US5586177A (en) | 1995-09-06 | 1996-12-17 | Bell Atlantic Network Services, Inc. | Intelligent signal transfer point (ISTP) |
US5583926A (en) | 1994-12-30 | 1996-12-10 | Stentor Resource Centre Inc. | Method and apparatus for routing a call to a number corresponding to a virtual public dial plan or to an existing dial plan |
US5884032A (en) * | 1995-09-25 | 1999-03-16 | The New Brunswick Telephone Company, Limited | System for coordinating communications via customer contact channel changing system using call centre for setting up the call between customer and an available help agent |
EP0867091B1 (en) * | 1995-12-11 | 2005-04-13 | Hewlett-Packard Company, A Delaware Corporation | Call setup gateway for telecommunications system |
CA2165856C (en) | 1995-12-21 | 2001-09-18 | R. William Carkner | Number portability with database query |
CA2165857C (en) | 1995-12-21 | 2000-07-25 | L. Lloyd Williams | Number portability using isup message option |
US5692033A (en) * | 1996-01-22 | 1997-11-25 | Bell Atlantic Network Services, Inc. | AIN queuing for call-back system |
CA2209238C (en) * | 1997-06-27 | 2000-07-25 | Bell Canada | Method and apparatus for monitoring selected telecommunications sessions in an intelligent switched telephone network |
-
1998
- 1998-05-01 US US09/071,092 patent/US6236722B1/en not_active Expired - Lifetime
-
1999
- 1999-05-03 EP EP99919006A patent/EP1075768B1/en not_active Expired - Lifetime
- 1999-05-03 AT AT99919006T patent/ATE231314T1/en not_active IP Right Cessation
- 1999-05-03 CA CA002270601A patent/CA2270601C/en not_active Expired - Lifetime
- 1999-05-03 WO PCT/CA1999/000398 patent/WO1999057915A1/en active IP Right Grant
- 1999-05-03 BR BR9910170-0A patent/BR9910170A/en not_active IP Right Cessation
- 1999-05-03 AU AU36961/99A patent/AU3696199A/en not_active Abandoned
- 1999-05-03 ES ES99919006T patent/ES2192381T3/en not_active Expired - Lifetime
- 1999-05-03 DE DE69904931T patent/DE69904931T2/en not_active Expired - Fee Related
Also Published As
Publication number | Publication date |
---|---|
DE69904931T2 (en) | 2004-02-19 |
ATE231314T1 (en) | 2003-02-15 |
ES2192381T3 (en) | 2003-10-01 |
AU3696199A (en) | 1999-11-23 |
CA2270601A1 (en) | 1999-11-01 |
DE69904931D1 (en) | 2003-02-20 |
EP1075768A1 (en) | 2001-02-14 |
CA2270601C (en) | 2002-05-28 |
US6236722B1 (en) | 2001-05-22 |
BR9910170A (en) | 2001-01-09 |
WO1999057915A1 (en) | 1999-11-11 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
EP1075768B1 (en) | Method and system for improved call setup | |
EP1042922B1 (en) | Method and system for completing a voice connection in a switched telephone network | |
US7436940B2 (en) | Methods and systems for enabling a reply call to voice mail message | |
US5692033A (en) | AIN queuing for call-back system | |
US5282243A (en) | Recording of automatic number identification to identify lost customers | |
US8670549B2 (en) | Method and system for improved routing of repair calls to a call center | |
US6055305A (en) | Method and apparatus for providing network-based customized call treatment | |
US6421437B1 (en) | System and method for re-directing incoming calls | |
US6301349B1 (en) | Method and system to connect an unanswered forwarded communication directly to a voice mail service | |
US6584178B2 (en) | Method and system for termination blocking of message delivery service in a switch-based telecommunication system | |
US8111816B2 (en) | Methods and systems for enabling return to same position in a review of messages in a voice mail system using tag or identifier stored in the voice mail system | |
US20100074429A1 (en) | System and Method for Forwarding Selective Calls | |
US6208723B1 (en) | System and method for enhanced automatic recall | |
US7945038B2 (en) | Methods and systems for releasing a voice mail system from a communication and further processing the communication | |
US6252954B1 (en) | System and method for delaying the ringing of a line | |
US6829342B2 (en) | System and method for handling voice calls and data calls | |
WO1999003301A2 (en) | System and method for providing a busy signal to a communication | |
MXPA00010509A (en) | Method and system for improved call setup | |
JP3376339B2 (en) | Call monitoring system using intelligent network system | |
CA2225937C (en) | Method and system for completing a voice connection between first and second voice terminals in a switched telephone network | |
CA2327912A1 (en) | Intelligent-networked telephone system with internet connection interruption |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
PUAI | Public reference made under article 153(3) epc to a published international application that has entered the european phase |
Free format text: ORIGINAL CODE: 0009012 |
|
17P | Request for examination filed |
Effective date: 20000927 |
|
AK | Designated contracting states |
Kind code of ref document: A1 Designated state(s): AT BE CH CY DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE |
|
GRAG | Despatch of communication of intention to grant |
Free format text: ORIGINAL CODE: EPIDOS AGRA |
|
17Q | First examination report despatched |
Effective date: 20010911 |
|
GRAG | Despatch of communication of intention to grant |
Free format text: ORIGINAL CODE: EPIDOS AGRA |
|
GRAG | Despatch of communication of intention to grant |
Free format text: ORIGINAL CODE: EPIDOS AGRA |
|
GRAH | Despatch of communication of intention to grant a patent |
Free format text: ORIGINAL CODE: EPIDOS IGRA |
|
GRAH | Despatch of communication of intention to grant a patent |
Free format text: ORIGINAL CODE: EPIDOS IGRA |
|
GRAA | (expected) grant |
Free format text: ORIGINAL CODE: 0009210 |
|
AK | Designated contracting states |
Kind code of ref document: B1 Designated state(s): AT BE CH CY DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: NL Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 Ref country code: LI Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 Ref country code: IT Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT;WARNING: LAPSES OF ITALIAN PATENTS WITH EFFECTIVE DATE BEFORE 2007 MAY HAVE OCCURRED AT ANY TIME BEFORE 2007. THE CORRECT EFFECTIVE DATE MAY BE DIFFERENT FROM THE ONE RECORDED. Effective date: 20030115 Ref country code: GR Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 Ref country code: FI Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 Ref country code: CH Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 Ref country code: BE Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 Ref country code: AT Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030115 |
|
REG | Reference to a national code |
Ref country code: GB Ref legal event code: FG4D Ref country code: CH Ref legal event code: EP |
|
REG | Reference to a national code |
Ref country code: IE Ref legal event code: FG4D |
|
REF | Corresponds to: |
Ref document number: 69904931 Country of ref document: DE Date of ref document: 20030220 Kind code of ref document: P |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: SE Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030415 Ref country code: PT Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030415 Ref country code: DK Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030415 |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: LU Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20030503 Ref country code: CY Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT Effective date: 20030503 |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: IE Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20030505 |
|
REG | Reference to a national code |
Ref country code: GB Ref legal event code: 732E |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: MC Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20030531 |
|
NLV1 | Nl: lapsed or annulled due to failure to fulfill the requirements of art. 29p and 29m of the patents act | ||
REG | Reference to a national code |
Ref country code: CH Ref legal event code: PL |
|
ET | Fr: translation filed | ||
REG | Reference to a national code |
Ref country code: ES Ref legal event code: FG2A Ref document number: 2192381 Country of ref document: ES Kind code of ref document: T3 |
|
PLBE | No opposition filed within time limit |
Free format text: ORIGINAL CODE: 0009261 |
|
STAA | Information on the status of an ep patent application or granted ep patent |
Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT |
|
REG | Reference to a national code |
Ref country code: ES Ref legal event code: PC2A |
|
26N | No opposition filed |
Effective date: 20031016 |
|
REG | Reference to a national code |
Ref country code: IE Ref legal event code: MM4A |
|
REG | Reference to a national code |
Ref country code: FR Ref legal event code: TP |
|
REG | Reference to a national code |
Ref country code: FR Ref legal event code: CD |
|
REG | Reference to a national code |
Ref country code: ES Ref legal event code: PC2A |
|
PGFP | Annual fee paid to national office [announced via postgrant information from national office to epo] |
Ref country code: ES Payment date: 20070524 Year of fee payment: 9 |
|
PGFP | Annual fee paid to national office [announced via postgrant information from national office to epo] |
Ref country code: DE Payment date: 20070726 Year of fee payment: 9 |
|
PGFP | Annual fee paid to national office [announced via postgrant information from national office to epo] |
Ref country code: GB Payment date: 20070521 Year of fee payment: 9 |
|
REG | Reference to a national code |
Ref country code: HK Ref legal event code: WD Ref document number: 1038668 Country of ref document: HK |
|
GBPC | Gb: european patent ceased through non-payment of renewal fee |
Effective date: 20080503 |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: DE Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20081202 |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: GB Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20080503 |
|
REG | Reference to a national code |
Ref country code: ES Ref legal event code: FD2A Effective date: 20080505 |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: ES Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20080505 |
|
PGFP | Annual fee paid to national office [announced via postgrant information from national office to epo] |
Ref country code: FR Payment date: 20091117 Year of fee payment: 11 |
|
REG | Reference to a national code |
Ref country code: FR Ref legal event code: ST Effective date: 20110131 |
|
PG25 | Lapsed in a contracting state [announced via postgrant information from national office to epo] |
Ref country code: FR Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES Effective date: 20100531 |